The invention relates to the general field of telecommunications. More particularly, the invention relates to managing quality of service during a call between a terminal connected to a home gateway and a remote terminal by using the session initiation protocol (SIP).
In known manner, a home gateway connected to a remote network enables a local terminal to set up a call with a remote terminal by using a voice over Internet protocol (VoIP). Typically, the local terminal uses the SIP protocol (as standardized in document RFC 3161) in order to set up the call, and in particular in order to negotiate with the remote terminal to determine which codec(s) are to be used.
By way of example, the local terminal comprises an SIP agent (known as a “user agent” (UA)) that is internal to the gateway, and that has an analog or a digital telephone connected thereto. In this context, the home gateway is capable of detecting the bandwidth of its access to the Internet and of suggesting to the SIP agent that it uses codecs that are compatible with the detected bandwidth.
Furthermore, users generally desire to make use of a plurality of local terminals connected to the gateway in order to make calls simultaneously. For example, a first terminal comprises an SIP agent that is internal to the gateway and connected to a digital or an analog telephone, whereas additional terminals are terminals that are external to the gateway, and as constituted by a multifunction mobile telephone or by a personal computer executing VOIP telephony software (“Softphone”), or indeed hardware terminals of the tablet, new generation TV, . . . , type. These external terminals may be connected to the gateway via a wireless connection, e.g. of the WiFi type, or via a wired connection, e.g. of the Ethernet type. Document FR 2 909 820 thus describes configuring a home gateway connected to four local terminals.
In such a situation, it is known to share the bandwidth of the access connection by allocating a virtual channel to each type of traffic. Thus, a conversational virtual channel is dedicated to the SIP agent internal to the gateway and an Internet virtual channel is dedicated to traffic coming from the external terminal. The conversational virtual channel is considered as having priority, and it is therefore dimensioned for a telephone conversation or for a video phone call, whereas the Internet virtual channel operates in the so-called “best effort” mode. Thus, if the conversational virtual channel is in use by the internal SIP agent for a conversation, then the bandwidth available for a telephone call coming from an external terminal may not be sufficient, given the selected codec, and conversation quality may be degraded.
It is also known to use a single virtual channel and to prioritize the various types of traffic by making use of IP packet marking and a plurality of packet processing queues, making prioritization available. Under such circumstances, if a priority queue is dedicated to the internal SIP agent and a best effort queue is dedicated to Internet traffic coming from the external terminals, the same problem arises as that described above, namely possible degradation of the quality of the conversation with the external terminal. If the same queue is used for the voice conversations coming from the internal SIP agent and from the external terminals, then the quality of all of the conversations can be degraded during simultaneous conversations.
There is thus a need for better management of the quality of service when a plurality of terminals make use of the same home gateway for simultaneous calls.